Bandwidth of a public packet switching network is shared by all kinds of applications to achieve statistical multiplexing gain. This share of resource introduces considerable uncertainty in workload and resource requirements. While delivering packets through such a network, the unpredictable delay jitter may introduce underflow or overflow in a limited client buffer even if the network is error-free at a time period. However, different from conventional text/image network applications, multimedia applications require end-to-end quality-of-service (QoS) with jitter-free playback of audio and video. Thus a good end-to-end flow control mechanism is needed to maintain high throughput and keeping average delay per packet at a reasonable level for such time-critical applications like multimedia communications.
Flow control is after congestion control, one of the major objectives of connection management in the area of reliable connection oriented data transmission. Congestion control prevents the network in a whole to be flooded. In contrast, the main goal of the flow control is to prevent a slow or even busy reception node from being flooded by a faster data sender. “Rate-based” and “window-based” mechanisms are the two best-known approaches for flow control.
General TCP flow control which is described in [Pos81] (Postel, Jon: Transmission Control Protocol—DARPA Internet Program Protocol Specification, RFC 793, University of Southern California—Information Sciences Institute (1981)) uses a window-based flow control mechanism. A receiver returns a “window” with every acknowledgment indicating a range of acceptable sequence numbers beyond the last received segment. The window indicates an allowed number of bytes the sender may transmit before receiving further permission. The disadvantage of this method is a burst way of sending, which even amplifies on high Bandwidth Delay Product links (also called “high BDP links”).
On the other hand, rate-based flow control can provide end-to-end deterministic and statistical performance guarantees over packet switching networks. The rate adjustment is performed by the intensive feedback controls from client to achieve guaranteed QoS. One rate-based approach is described in [Wan01] (Wang, Chia-Hui; Ho, Jan-Ming; Chang, Ray-I and Hsu, Shun-Chin: A Control-Theoretic Mechanism for Rate-based Flow Control of Realtime Multimedia Communication, Techn. Ber., Department of Computer Science and Information Engineering, National Taiwan University, Taipei, Taiwan, R.O.C. and Institute of Information Science, Academia Sinica, Taipei, Taiwan, R.O.C. (2001), multimedia, Internet, Video Technologies 2001 (MIV 2001) of WSES/IEEE International Multi-conference). This method uses a closed loop control mechanism to adapt the sender's data rate. The control theoretic “device to control” is the receive-buffer-occupancy (BO). If the BO value exceeds a given threshold, the receiver gives a feedback about its BO to the sender. The sender uses a Proportional, Derivative (PD) rate control to calculate a new sending rate from the received BO feedback. The method proposed by Wang relies only on the current sending rate and the changes of the receive-buffer-occupancy. Effective receive rate of an application located on receiver is not directly taken into account. That is the reason why fluctuations of this rate stay almost unaccounted. On high BDP links this could lead to unintended buffer overflows or underruns, respectively. A drawback of method proposed by Wang is that it ignores also amount of data currently readable from buffer, which could cause unnecessary or even wrong speed adjustments.
One of the goals of the present invention is to propose a rate-based flow control mechanism providing a more effective and reliable data flow control even on erroneous high BDP links than state-of-the art flow Control mechanisms and allowing a more efficient network resources utilization than the known approaches.